Countless excellent, expensive and beloved audio systems comprising conventional amplifiers and passive loudspeakers are installed all around in living rooms, listening rooms, home cinemas, conference rooms, concert halls, studios, etc., or are set up, packed, moved, set up, etc., by public address companies, band crews, etc. Such systems do typically not provide any means for obtaining information about the acoustical or spatial properties of the setup or surroundings. Other systems for obtaining such information have been provided, but require typically that separate measure microphones are set up, the speakers exchanged with self-calibrating active speakers or active or passive speakers comprising separate measure microphones installed, etc. Hence, no simple, automatic or semi-automatic means exists for the numerous owners of passive loudspeaker audio systems to obtain such information, if they want to keep using their existing loudspeakers and amplifiers.
The perceived sound quality of loudspeakers is affected by the listening room in several ways, typically referred to as boundary effect, room modes, discrete reflections and reverberation.
By boundary effect is referred to a particular type of interference that may occur for low frequency audio when a speaker is placed near walls or other reflective surfaces, as the direct sound from the loudspeaker is superposed with the sound reflected from the surfaces. The reflected, sounds appear to emanate from “mirror image sources” that are the physical speaker's geometrical mirror images in the surfaces. At very low frequencies, where the acoustical wavelength is many meters, e.g. 11.4 meters at 30 Hz, the direct sound and the reflections add up in constructive interference, because the differences in propagation distance from each source, mirror image source or real source, to listening position are much smaller than the wavelength. In this situation a 6 dB increase, i.e. a doubling of sound pressure, can be observed with every surface added, so a speaker placed in a corner, i.e. 3 boundaries, produces up to 18 dB more very-low-frequency sound pressure level at listening position than it would have in open air at the same distance. By sound pressure level is referred to
  SPL  =      20    ⁢                  log        10            ⁡              (                              p            RMS                                              20              ·                              10                                  -                  6                                                      ⁢                                                  ⁢            Pa                          )            where pRMS is the sound pressure in Pascal, and SPL is measured in decibels, dB. With decreasing wavelength, i.e. increasing frequency, the interference pattern becomes more complex with varying combinations of constructive and destructive interference between direct sound and reflections. This amounts to a significant deviation from a neutral, flat low-frequency response, and the deviation pattern is highly dependent on speaker placement with respect to the 3 nearest boundaries, e.g. floor, rear wall, side wall, and also dependent on surface absorption properties. This room-dependent low-mid-frequency coloration is called the boundary effect. Some consumer loudspeakers come with specific positioning recommendations and some even with built-in rudimentary equalization means for compensating the boundary effect, but in reality the boundary effect remains a great source of uncertainty in achieving a neutral reproduction of speech and music from quality loudspeakers. However the degrading influence of the boundary effect on sound reproduction can be greatly reduced by suitable equalization, that is: Filtering of the audio signal before it is sent to the speakers. A problem related to this is, however, how to determine the equalization parameters that may cause a reduction of the boundary effect without adding further or alternative degradation to the sound production.
Room modes refer to a different type of interference that occurs in closed rooms. In a closed room, the propagation path of higher-order reflections (reflections of reflections of reflections of . . . ) can form closed loops, the simplest case being the “ping-pong” propagation of a reflecting sound between two parallel walls. At frequencies where the propagation distance through one cycle of the loop is an integral number of wavelengths, all “generations” of the looped sound propagation are in phase, and a self-reinforcing, geometrically fixed pattern of sound is established in the room, with high sound pressure accumulating at certain places near the surfaces (particularly in corners where more surfaces meet) and high particle velocity (but low pressure) accumulating at other places in mid-air. For box-shaped rooms, this condition is fulfilled at frequencies
      f          x      ,      y      ,      z        =            c      2        ⁢                                        (                                          n                x                                            l                x                                      )                    2                +                              (                                          n                y                                            l                y                                      )                    2                +                              (                                          n                z                                            l                z                                      )                    2                    where lxyz are room dimensions, nxyz are non-negative integers and c is the speed of sound. The particle velocity in and out of the room surfaces is of course minimal, actually zero for an ideal reflector. Such a pattern in called a room mode. In normal rooms, the SPL at pressure maxima can easily be 20 dB above average. This severe coloration is dependent on both listening position and speaker position. The mode acts as an imperfect energy accumulator and the speaker's ability to charge power into the “accumulator” depends strongly on its positioning within the geometrical modal pattern. Normal direct-radiating loudspeakers produce nearly constant volume-velocity, irrespective of the sound pressure on the speaker surface; hence, they inject maximal power into the mode when placed at pressure maxima, typically in a corner. Besides causing wild fluctuations in the steady-state frequency response that depend on both speaker and listening positions, the accumulating effect of the modes also provides the room with memory. The charging of the “accumulator” takes time, and when the source sound is cut off, the “accumulator” discharges through sound absorption. This memory effect is clearly demonstrable if for instance the door of a room is slammed and the decay of the sound observed, especially if the decaying sound is observed from a room corner. The room superposes the same tonal decay on the music played by loudspeakers. Thus, the room modes create highly frequency-dependent time smearing which also shows as peaks in the effective decay time of the room as a function of frequency. The decay time T60 is the time it takes to decay 60 dB and is determined by the room volume Vroom and the combined equivalent absorption area of the room surfaces Si with their absorption coefficients αi:
            T      60        ⁡          (      f      )        =                              V          room                                      ∑                          i              =              1                                      N              materials                                ⁢                                    S              i                        ⁢                                          α                i                            ⁡                              (                f                )                                                        ·      0.161        ⁢                  ⁢          m              -        1              ⁢    s  
As mentioned, the room modes' effect on the (steady-state) frequency response of the audio reproduction system is highly position dependent. Therefore, equalization can only cure this problem at one or maybe a few selected listening positions. Added low-frequency absorption, in the form of passive absorbers or auxiliary subwoofers acting as active absorbers, appears to be the only overall cure for room modes. The time-smearing problem can be solved by modal equalization, but this requires a delicate identification of each separate room mode's frequency and damping. Modal equalization comprises cancelling the frequency domain poles of the room with zeros and placing new poles electronically at the same frequencies, but with damping factors corresponding to the room's overall low-frequency decay time. Such methods have been described further in the documents Makivirta, Karjalainen et al.: “Low-Frequency Modal Equalization Of Loudspeaker-Room Responses”, AES Convention Paper 5480, hereby incorporated by reference, Karjalainen et al.: “Estimation of Modal Decay Parameters from Noisy Response Measurements”, JAES Vol. 50 No. 11, November 2002, hereby incorporated by reference, Karjalainen et al.: “Frequency-Zooming ARMA Modeling of Resonant and Reverberant Systems”, JAES Vol. 50 No. 12, December 2002, hereby incorporated by reference, and Rhonda J Wilson et al.: “The Loudspeaker-Room Interface—Controlling Excitation of Room Modes”, Presented at 23rd International AES Conference, Copenhagen, Denmark, May 23-25, 2003, hereby incorporated by reference. A problem related to these methods is, however, how to determine the room modes, and thereby the poles to cancel.
Regarding discrete reflection at mid-to-high frequencies, reflections from room boundaries are more likely to be absorbed or diffused. If they are not, and this causes audible disturbance, there is very little to do about it in terms of signal processing. Adding passive absorption to the room becomes a much more feasible option at the shorter wavelengths. Carpets and curtains or even quite thin panels of absorbent material will generally do the job.
Border zone cases between boundary effect and discrete reflections are floor/ceiling reflections in domestic setups and console reflections in studio monitoring. Here the reflection arrives from the same azimuth angle as the direct sound, causing near-identical comb-filtering of the signals reaching both the listener's ears. Therefore, if this problem is not prevented from the outset by controlled vertical speaker directivity, equalization may still help. A problem related to this is, however, how to determine the equalization parameters that may cause such help.
The reverberant sound field is the semi-random (diffuse) mixture of all the higher-order reflections in the room. Unlike the modes, this does not add up in phase, hence the randomness. Ideally the diffuse sound field has no direction of propagation (i.e. no non-zero intensity vector) at any point. It is characterized by statistical means, namely the decay time. When the sound source is turned off, the diffuse sound field decays exponentially due to absorption in room surfaces and air.
As mentioned earlier, the decay time is a function of frequency f. If the decay time is too long in any part of the spectrum, degrading speech intelligibility and/or cluttering up the sound image in the recording, the only cures are adding absorption to the room or applying more directive loudspeakers, reducing the injection of sound power into the reverberant field. If the spectral color of the reverberation is too bright or too dull compared to what the loudspeaker manufacturer and record producer anticipated, a gentle, smoothly sloping; “tilt” equalizing filter may help, even though this will also affect the direct sound. If the reverberant sound field in the room is not sufficiently diffuse, diffusers (passive or active) can be added to the room. Finally, if the room is too “dry” (decay time too low), artificial reverberation can be added by running the audio signal through a suitable reverb algorithm and/or by installing an active room enhancement system, i.e. a complex network of reverb algorithms, amplifiers and loudspeakers, sometimes with microphones placed in the same room contributing to the network input. A problem related to improving the reverberation is how to automatically determine the way the current loudspeaker setup couples to the current room, in order to automatically suggest or perform a suitable equalization.
Existing automatic room correction systems on the market can be divided into systems with user-operated test microphones and systems with self-calibrating speakers.
The systems with user-operated test microphones are far the dominant class on the market. The reasoning is clear and logical: The sound that is heard must be measured before it can be improved. Usually this involves a measurement of the frequency response or the impulse response (may be obtained by two-channel analysis with any broad-band test signal) from each amplifier channel (voltage) to sound pressure at one or more target positions in the listening area. These measurements are then analyzed and transformed into an equalizer target response according to the chosen equalization philosophy (method). The equalization filter may then be automatically implemented in a DSP program. The test microphone is normally omni-directional (pressure sensitive), but some equalization philosophies may require other microphone types, such as cardioid or sound-field microphones. Within this very broad class of systems, any acoustical properties of room and loudspeakers can be measured and dealt with according to the preferred equalization philosophy. These systems and methods, however, require the user to obtain measurement equipment, perform time-consuming and cumbersome measurements according to advanced measuring schemes, and, for perfect results, do this anytime the listening position or room is changed, e.g. replacement or movement of furniture, speakers, listening position(s), etc. Furthermore, it may for some systems be a complex task to determine and implement equalization parameters suitable for reducing degradation of sound quality originating from the measured speaker-room coupling.
Of self-calibrating speaker systems the major system is Bang & Olufsen's Adaptive Bass Control (ABC), e.g.: available in the flagship product Beolab 5. The ABC technique is disclosed in European patents EP 0 772 374 and EP 1 133 896. The system employs a moving microphone for measuring the speaker's sound pressure responses and the sound pressure gradient responses very near the speaker itself. From this the acoustical radiation resistance presented to the speaker by the room and the speaker's acoustical power response (which is essentially proportional to the radiation resistance) in the actual position and environment are derived and transformed into an equalizer target response. This equalization philosophy, which is applied in the frequency range below 500 Hz, takes excellent care of the boundary effect problem. However, these intelligent speakers don't know anything about the listening position. So even though a speaker placement in a modal pressure maximum will be detectable, they are not able to know if the detected mode will result in a frequency response peak at listening position or not. A self-calibrating speaker system like the ABC does however require the user to replace his conventional speakers with the self-calibrating speakers, which are so far extremely expensive, and only available in very few configurations.
It is an object of the present invention to provide a method and system for performing acoustical measurements by means of an audio system comprising passive loudspeakers, and thereby facilitate owners of such systems to obtain acoustical and/or spatial information without exchanging their equipment.
It is a further object of the present invention to provide a method and system for automatically determining properties of the couplings between conventional, passive speakers and the listening room.
It is a further object of the present invention to provide a method and system for establishing and implementing equalization parameters suitable for correcting the determined couplings.